dlls/dsound/mixer.c

   1 /*                      DirectSound
   2  *
   3  * Copyright 1998 Marcus Meissner
   4  * Copyright 1998 Rob Riggs
   5  * Copyright 2000-2002 TransGaming Technologies, Inc.
   6  * Copyright 2007 Peter Dons Tychsen
   7  * Copyright 2007 Maarten Lankhorst
   8  * Copyright 2011 Owen Rudge for CodeWeavers
   9  *
  10  * This library is free software; you can redistribute it and/or
  11  * modify it under the terms of the GNU Lesser General Public
  12  * License as published by the Free Software Foundation; either
  13  * version 2.1 of the License, or (at your option) any later version.
  14  *
  15  * This library is distributed in the hope that it will be useful,
  16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  18  * Lesser General Public License for more details.
  19  *
  20  * You should have received a copy of the GNU Lesser General Public
  21  * License along with this library; if not, write to the Free Software
  22  * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
  23  */
  24
  25 #include <assert.h>
  26 #include <stdarg.h>
  27 #include <math.h>       /* Insomnia - pow() function */
  28
  29 #define COBJMACROS
  30
  31 #include "windef.h"
  32 #include "winbase.h"
  33 #include "mmsystem.h"
  34 #include "wingdi.h"
  35 #include "mmreg.h"
  36 #include "winternl.h"
  37 #include "wine/debug.h"
  38 #include "dsound.h"
  39 #include "ks.h"
  40 #include "ksmedia.h"
  41 #include "dsound_private.h"
  42 #include "fir.h"
  43
  44 WINE_DEFAULT_DEBUG_CHANNEL(dsound);
  45
  46 void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
  47 {
  48         double temp;
  49         TRACE("(%p)\n",volpan);
  50
  51         TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  52         /* the AmpFactors are expressed in 16.16 fixed point */
  53
  54         /* FIXME: use calculated vol and pan ampfactors */
  55         temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
  56         volpan->dwTotalAmpFactor[0] = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  57         temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
  58         volpan->dwTotalAmpFactor[1] = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
  59
  60         TRACE("left = %x, right = %x\n", volpan->dwTotalAmpFactor[0], volpan->dwTotalAmpFactor[1]);
  61 }
  62
  63 void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
  64 {
  65     double left,right;
  66     TRACE("(%p)\n",volpan);
  67
  68     TRACE("left=%x, right=%x\n",volpan->dwTotalAmpFactor[0],volpan->dwTotalAmpFactor[1]);
  69     if (volpan->dwTotalAmpFactor[0]==0)
  70         left=-10000;
  71     else
  72         left=600 * log(((double)volpan->dwTotalAmpFactor[0]) / 0xffff) / log(2);
  73     if (volpan->dwTotalAmpFactor[1]==0)
  74         right=-10000;
  75     else
  76         right=600 * log(((double)volpan->dwTotalAmpFactor[1]) / 0xffff) / log(2);
  77     if (left<right)
  78         volpan->lVolume=right;
  79     else
  80         volpan->lVolume=left;
  81     if (volpan->lVolume < -10000)
  82         volpan->lVolume=-10000;
  83     volpan->lPan=right-left;
  84     if (volpan->lPan < -10000)
  85         volpan->lPan=-10000;
  86
  87     TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
  88 }
  89
  90 /**
  91  * Recalculate the size for temporary buffer, and new writelead
  92  * Should be called when one of the following things occur:
  93  * - Primary buffer format is changed
  94  * - This buffer format (frequency) is changed
  95  */
  96 void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
  97 {
  98         DWORD ichannels = dsb->pwfx->nChannels;
  99         DWORD ochannels = dsb->device->pwfx->nChannels;
 100         WAVEFORMATEXTENSIBLE *pwfxe;
 101         BOOL ieee = FALSE;
 102
 103         TRACE("(%p)\n",dsb);
 104
 105         pwfxe = (WAVEFORMATEXTENSIBLE *) dsb->pwfx;
 106         dsb->freqAdjustNum = dsb->freq;
 107         dsb->freqAdjustDen = dsb->device->pwfx->nSamplesPerSec;
 108
 109         if ((pwfxe->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) || ((pwfxe->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE)
 110             && (IsEqualGUID(&pwfxe->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))))
 111                 ieee = TRUE;
 112
 113         /**
 114          * Recalculate FIR step and gain.
 115          *
 116          * firstep says how many points of the FIR exist per one
 117          * sample in the secondary buffer. firgain specifies what
 118          * to multiply the FIR output by in order to attenuate it correctly.
 119          */
 120         if (dsb->freqAdjustNum / dsb->freqAdjustDen > 0) {
 121                 /**
 122                  * Yes, round it a bit to make sure that the
 123                  * linear interpolation factor never changes.
 124                  */
 125                 dsb->firstep = fir_step * dsb->freqAdjustDen / dsb->freqAdjustNum;
 126         } else {
 127                 dsb->firstep = fir_step;
 128         }
 129         dsb->firgain = (float)dsb->firstep / fir_step;
 130
 131         /* calculate the 10ms write lead */
 132         dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
 133
 134         dsb->freqAccNum = 0;
 135
 136         dsb->get_aux = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
 137         dsb->put_aux = putieee32;
 138
 139         dsb->get = dsb->get_aux;
 140         dsb->put = dsb->put_aux;
 141
 142         if (ichannels == ochannels)
 143         {
 144                 dsb->mix_channels = ichannels;
 145                 if (ichannels > 32) {
 146                         FIXME("Copying %u channels is unsupported, limiting to first 32\n", ichannels);
 147                         dsb->mix_channels = 32;
 148                 }
 149         }
 150         else if (ichannels == 1)
 151         {
 152                 dsb->mix_channels = 1;
 153
 154                 if (ochannels == 2)
 155                         dsb->put = put_mono2stereo;
 156                 else if (ochannels == 4)
 157                         dsb->put = put_mono2quad;
 158                 else if (ochannels == 6)
 159                         dsb->put = put_mono2surround51;
 160         }
 161         else if (ochannels == 1)
 162         {
 163                 dsb->mix_channels = 1;
 164                 dsb->get = get_mono;
 165         }
 166         else if (ichannels == 2 && ochannels == 4)
 167         {
 168                 dsb->mix_channels = 2;
 169                 dsb->put = put_stereo2quad;
 170         }
 171         else if (ichannels == 2 && ochannels == 6)
 172         {
 173                 dsb->mix_channels = 2;
 174                 dsb->put = put_stereo2surround51;
 175         }
 176         else if (ichannels == 6 && ochannels == 2)
 177         {
 178                 dsb->mix_channels = 6;
 179                 dsb->put = put_surround512stereo;
 180                 dsb->put_aux = putieee32_sum;
 181         }
 182         else if (ichannels == 4 && ochannels == 2)
 183         {
 184                 dsb->mix_channels = 4;
 185                 dsb->put = put_quad2stereo;
 186                 dsb->put_aux = putieee32_sum;
 187         }
 188         else
 189         {
 190                 if (ichannels > 2)
 191                         FIXME("Conversion from %u to %u channels is not implemented, falling back to stereo\n", ichannels, ochannels);
 192                 dsb->mix_channels = 2;
 193         }
 194 }
 195
 196 /**
 197  * Check for application callback requests for when the play position
 198  * reaches certain points.
 199  *
 200  * The offsets that will be triggered will be those between the recorded
 201  * "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
 202  * beyond that position.
 203  */
 204 void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len)
 205 {
 206     int first, left, right, check;
 207
 208     if(dsb->nrofnotifies == 0)
 209         return;
 210
 211     if(dsb->state == STATE_STOPPED){
 212         TRACE("Stopped...\n");
 213         /* DSBPN_OFFSETSTOP notifies are always at the start of the sorted array */
 214         for(left = 0; left < dsb->nrofnotifies; ++left){
 215             if(dsb->notifies[left].dwOffset != DSBPN_OFFSETSTOP)
 216                 break;
 217
 218             TRACE("Signalling %p\n", dsb->notifies[left].hEventNotify);
 219             SetEvent(dsb->notifies[left].hEventNotify);
 220         }
 221         return;
 222     }
 223
 224     for(first = 0; first < dsb->nrofnotifies && dsb->notifies[first].dwOffset == DSBPN_OFFSETSTOP; ++first)
 225         ;
 226
 227     if(first == dsb->nrofnotifies)
 228         return;
 229
 230     check = left = first;
 231     right = dsb->nrofnotifies - 1;
 232
 233     /* find leftmost notify that is greater than playpos */
 234     while(left != right){
 235         check = left + (right - left) / 2;
 236         if(dsb->notifies[check].dwOffset < playpos)
 237             left = check + 1;
 238         else if(dsb->notifies[check].dwOffset > playpos)
 239             right = check;
 240         else{
 241             left = check;
 242             break;
 243         }
 244     }
 245
 246     TRACE("Not stopped: first notify: %u (%u), left notify: %u (%u), range: [%u,%u)\n",
 247             first, dsb->notifies[first].dwOffset,
 248             left, dsb->notifies[left].dwOffset,
 249             playpos, (playpos + len) % dsb->buflen);
 250
 251     /* send notifications in range */
 252     if(dsb->notifies[left].dwOffset >= playpos){
 253         for(check = left; check < dsb->nrofnotifies; ++check){
 254             if(dsb->notifies[check].dwOffset >= playpos + len)
 255                 break;
 256
 257             TRACE("Signalling %p (%u)\n", dsb->notifies[check].hEventNotify, dsb->notifies[check].dwOffset);
 258             SetEvent(dsb->notifies[check].hEventNotify);
 259         }
 260     }
 261
 262     if(playpos + len > dsb->buflen){
 263         for(check = first; check < left; ++check){
 264             if(dsb->notifies[check].dwOffset >= (playpos + len) % dsb->buflen)
 265                 break;
 266
 267             TRACE("Signalling %p (%u)\n", dsb->notifies[check].hEventNotify, dsb->notifies[check].dwOffset);
 268             SetEvent(dsb->notifies[check].hEventNotify);
 269         }
 270     }
 271 }
 272
 273 static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
 274         DWORD mixpos, DWORD channel)
 275 {
 276     if (mixpos >= dsb->buflen && !(dsb->playflags & DSBPLAY_LOOPING))
 277         return 0.0f;
 278     return dsb->get(dsb, mixpos % dsb->buflen, channel);
 279 }
 280
 281 static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb, UINT count)
 282 {
 283     UINT istride = dsb->pwfx->nBlockAlign;
 284     UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
 285     DWORD channel, i;
 286     for (i = 0; i < count; i++)
 287         for (channel = 0; channel < dsb->mix_channels; channel++)
 288             dsb->put(dsb, i * ostride, channel, get_current_sample(dsb,
 289                     dsb->sec_mixpos + i * istride, channel));
 290     return count;
 291 }
 292
 293 static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNum)
 294 {
 295     UINT i, channel;
 296     UINT istride = dsb->pwfx->nBlockAlign;
 297     UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
 298
 299     LONG64 freqAcc_start = *freqAccNum;
 300     LONG64 freqAcc_end = freqAcc_start + count * dsb->freqAdjustNum;
 301     UINT dsbfirstep = dsb->firstep;
 302     UINT channels = dsb->mix_channels;
 303     UINT max_ipos = (freqAcc_start + count * dsb->freqAdjustNum) / dsb->freqAdjustDen;
 304
 305     UINT fir_cachesize = (fir_len + dsbfirstep - 2) / dsbfirstep;
 306     UINT required_input = max_ipos + fir_cachesize;
 307     float *intermediate, *fir_copy, *itmp;
 308
 309     DWORD len = required_input * channels;
 310     len += fir_cachesize;
 311     len *= sizeof(float);
 312
 313     if (!dsb->device->cp_buffer) {
 314         dsb->device->cp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
 315         dsb->device->cp_buffer_len = len;
 316     } else if (len > dsb->device->cp_buffer_len) {
 317         dsb->device->cp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->cp_buffer, len);
 318         dsb->device->cp_buffer_len = len;
 319     }
 320
 321     fir_copy = dsb->device->cp_buffer;
 322     intermediate = fir_copy + fir_cachesize;
 323
 324
 325     /* Important: this buffer MUST be non-interleaved
 326      * if you want -msse3 to have any effect.
 327      * This is good for CPU cache effects, too.
 328      */
 329     itmp = intermediate;
 330     for (channel = 0; channel < channels; channel++)
 331         for (i = 0; i < required_input; i++)
 332             *(itmp++) = get_current_sample(dsb,
 333                     dsb->sec_mixpos + i * istride, channel);
 334
 335     for(i = 0; i < count; ++i) {
 336         UINT int_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / dsb->freqAdjustDen;
 337         float total_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / (float)dsb->freqAdjustDen;
 338         UINT ipos = int_fir_steps / dsbfirstep;
 339
 340         UINT idx = (ipos + 1) * dsbfirstep - int_fir_steps - 1;
 341         float rem = int_fir_steps + 1.0 - total_fir_steps;
 342
 343         int fir_used = 0;
 344         while (idx < fir_len - 1) {
 345             fir_copy[fir_used++] = fir[idx] * (1.0 - rem) + fir[idx + 1] * rem;
 346             idx += dsb->firstep;
 347         }
 348
 349         assert(fir_used <= fir_cachesize);
 350         assert(ipos + fir_used <= required_input);
 351
 352         for (channel = 0; channel < dsb->mix_channels; channel++) {
 353             int j;
 354             float sum = 0.0;
 355             float* cache = &intermediate[channel * required_input + ipos];
 356             for (j = 0; j < fir_used; j++)
 357                 sum += fir_copy[j] * cache[j];
 358             dsb->put(dsb, i * ostride, channel, sum * dsb->firgain);
 359         }
 360     }
 361
 362     *freqAccNum = freqAcc_end % dsb->freqAdjustDen;
 363
 364     return max_ipos;
 365 }
 366
 367 static void cp_fields(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNum)
 368 {
 369     DWORD ipos, adv;
 370
 371     if (dsb->freqAdjustNum == dsb->freqAdjustDen)
 372         adv = cp_fields_noresample(dsb, count); /* *freqAccNum is unmodified */
 373     else
 374         adv = cp_fields_resample(dsb, count, freqAccNum);
 375
 376     ipos = dsb->sec_mixpos + adv * dsb->pwfx->nBlockAlign;
 377     if (ipos >= dsb->buflen) {
 378         if (dsb->playflags & DSBPLAY_LOOPING)
 379             ipos %= dsb->buflen;
 380         else {
 381             ipos = 0;
 382             dsb->state = STATE_STOPPED;
 383         }
 384     }
 385
 386     dsb->sec_mixpos = ipos;
 387 }
 388
 389 /**
 390  * Calculate the distance between two buffer offsets, taking wraparound
 391  * into account.
 392  */
 393 static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
 394 {
 395 /* If these asserts fail, the problem is not here, but in the underlying code */
 396         assert(ptr1 < buflen);
 397         assert(ptr2 < buflen);
 398         if (ptr1 >= ptr2) {
 399                 return ptr1 - ptr2;
 400         } else {
 401                 return buflen + ptr1 - ptr2;
 402         }
 403 }
 404 /**
 405  * Mix at most the given amount of data into the allocated temporary buffer
 406  * of the given secondary buffer, starting from the dsb's first currently
 407  * unsampled frame (writepos), translating frequency (pitch), stereo/mono
 408  * and bits-per-sample so that it is ideal for the primary buffer.
 409  * Doesn't perform any mixing - this is a straight copy/convert operation.
 410  *
 411  * dsb = the secondary buffer
 412  * writepos = Starting position of changed buffer
 413  * len = number of bytes to resample from writepos
 414  *
 415  * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
 416  */
 417 static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD frames)
 418 {
 419         UINT size_bytes = frames * sizeof(float) * dsb->device->pwfx->nChannels;
 420         HRESULT hr;
 421         int i;
 422
 423         if (dsb->device->tmp_buffer_len < size_bytes || !dsb->device->tmp_buffer)
 424         {
 425                 dsb->device->tmp_buffer_len = size_bytes;
 426                 if (dsb->device->tmp_buffer)
 427                         dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, size_bytes);
 428                 else
 429                         dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, size_bytes);
 430         }
 431         if(dsb->put_aux == putieee32_sum)
 432                 memset(dsb->device->tmp_buffer, 0, dsb->device->tmp_buffer_len);
 433
 434         cp_fields(dsb, frames, &dsb->freqAccNum);
 435
 436         if (size_bytes > 0) {
 437                 for (i = 0; i < dsb->num_filters; i++) {
 438                         if (dsb->filters[i].inplace) {
 439                                 hr = IMediaObjectInPlace_Process(dsb->filters[i].inplace, size_bytes, (BYTE*)dsb->device->tmp_buffer, 0, DMO_INPLACE_NORMAL);
 440
 441                                 if (FAILED(hr))
 442                                         WARN("IMediaObjectInPlace_Process failed for filter %u\n", i);
 443                         } else
 444                                 WARN("filter %u has no inplace object - unsupported\n", i);
 445                 }
 446         }
 447 }
 448
 449 static void DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT frames)
 450 {
 451         INT     i;
 452         float vols[DS_MAX_CHANNELS];
 453         UINT channels = dsb->device->pwfx->nChannels, chan;
 454
 455         TRACE("(%p,%d)\n",dsb,frames);
 456         TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalAmpFactor[0],
 457                 dsb->volpan.dwTotalAmpFactor[1]);
 458
 459         if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
 460             (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
 461              !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
 462                 return; /* Nothing to do */
 463
 464         if (channels > DS_MAX_CHANNELS)
 465         {
 466                 FIXME("There is no support for %u channels\n", channels);
 467                 return;
 468         }
 469
 470         for (i = 0; i < channels; ++i)
 471                 vols[i] = dsb->volpan.dwTotalAmpFactor[i] / ((float)0xFFFF);
 472
 473         for(i = 0; i < frames; ++i){
 474                 for(chan = 0; chan < channels; ++chan){
 475                         dsb->device->tmp_buffer[i * channels + chan] *= vols[chan];
 476                 }
 477         }
 478 }
 479
 480 /**
 481  * Mix (at most) the given number of bytes into the given position of the
 482  * device buffer, from the secondary buffer "dsb" (starting at the current
 483  * mix position for that buffer).
 484  *
 485  * Returns the number of bytes actually mixed into the device buffer. This
 486  * will match fraglen unless the end of the secondary buffer is reached
 487  * (and it is not looping).
 488  *
 489  * dsb  = the secondary buffer to mix from
 490  * fraglen = number of bytes to mix
 491  */
 492 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, float *mix_buffer, DWORD frames)
 493 {
 494         float *ibuf;
 495         DWORD oldpos;
 496
 497         TRACE("sec_mixpos=%d/%d\n", dsb->sec_mixpos, dsb->buflen);
 498         TRACE("(%p, frames=%d)\n",dsb,frames);
 499
 500         /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
 501         oldpos = dsb->sec_mixpos;
 502         DSOUND_MixToTemporary(dsb, frames);
 503         ibuf = dsb->device->tmp_buffer;
 504
 505         /* Apply volume if needed */
 506         DSOUND_MixerVol(dsb, frames);
 507
 508         mixieee32(ibuf, mix_buffer, frames * dsb->device->pwfx->nChannels);
 509
 510         /* check for notification positions */
 511         if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
 512             dsb->state != STATE_STARTING) {
 513                 INT ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
 514                 DSOUND_CheckEvent(dsb, oldpos, ilen);
 515         }
 516
 517         return frames;
 518 }
 519
 520 /**
 521  * Mix some frames from the given secondary buffer "dsb" into the device
 522  * primary buffer.
 523  *
 524  * dsb = the secondary buffer
 525  * playpos = the current play position in the device buffer (primary buffer)
 526  * frames = the maximum number of frames in the primary buffer to mix, from the
 527  *          current writepos.
 528  *
 529  * Returns: the number of frames beyond the writepos that were mixed.
 530  */
 531 static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, float *mix_buffer, DWORD frames)
 532 {
 533         DWORD primary_done = 0;
 534
 535         TRACE("(%p, frames=%d)\n",dsb,frames);
 536         TRACE("looping=%d, leadin=%d\n", dsb->playflags, dsb->leadin);
 537
 538         /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
 539         /* FIXME: Is this needed? */
 540         if (dsb->leadin && dsb->state == STATE_STARTING) {
 541                 if (frames > 2 * dsb->device->frag_frames) {
 542                         primary_done = frames - 2 * dsb->device->frag_frames;
 543                         frames = 2 * dsb->device->frag_frames;
 544                         dsb->sec_mixpos += primary_done *
 545                                 dsb->pwfx->nBlockAlign * dsb->freqAdjustNum / dsb->freqAdjustDen;
 546                 }
 547         }
 548
 549         dsb->leadin = FALSE;
 550
 551         TRACE("frames (primary) = %i\n", frames);
 552
 553         /* First try to mix to the end of the buffer if possible
 554          * Theoretically it would allow for better optimization
 555         */
 556         primary_done += DSOUND_MixInBuffer(dsb, mix_buffer, frames);
 557
 558         TRACE("total mixed data=%d\n", primary_done);
 559
 560         /* Report back the total prebuffered amount for this buffer */
 561         return primary_done;
 562 }
 563
 564 /**
 565  * For a DirectSoundDevice, go through all the currently playing buffers and
 566  * mix them in to the device buffer.
 567  *
 568  * frames = the maximum amount to mix into the primary buffer
 569  * all_stopped = reports back if all buffers have stopped
 570  *
 571  * Returns:  the length beyond the writepos that was mixed to.
 572  */
 573
 574 static void DSOUND_MixToPrimary(const DirectSoundDevice *device, float *mix_buffer, DWORD frames, BOOL *all_stopped)
 575 {
 576         INT i;
 577         IDirectSoundBufferImpl  *dsb;
 578
 579         /* unless we find a running buffer, all have stopped */
 580         *all_stopped = TRUE;
 581
 582         TRACE("(frames %d)\n", frames);
 583         for (i = 0; i < device->nrofbuffers; i++) {
 584                 dsb = device->buffers[i];
 585
 586                 TRACE("MixToPrimary for %p, state=%d\n", dsb, dsb->state);
 587
 588                 if (dsb->buflen && dsb->state) {
 589                         TRACE("Checking %p, frames=%d\n", dsb, frames);
 590                         RtlAcquireResourceShared(&dsb->lock, TRUE);
 591                         /* if buffer is stopping it is stopped now */
 592                         if (dsb->state == STATE_STOPPING) {
 593                                 dsb->state = STATE_STOPPED;
 594                                 DSOUND_CheckEvent(dsb, 0, 0);
 595                         } else if (dsb->state != STATE_STOPPED) {
 596
 597                                 /* if the buffer was starting, it must be playing now */
 598                                 if (dsb->state == STATE_STARTING)
 599                                         dsb->state = STATE_PLAYING;
 600
 601                                 /* mix next buffer into the main buffer */
 602                                 DSOUND_MixOne(dsb, mix_buffer, frames);
 603
 604                                 *all_stopped = FALSE;
 605                         }
 606                         RtlReleaseResource(&dsb->lock);
 607                 }
 608         }
 609 }
 610
 611 /**
 612  * Add buffers to the emulated wave device system.
 613  *
 614  * device = The current dsound playback device
 615  * force = If TRUE, the function will buffer up as many frags as possible,
 616  *         even though and will ignore the actual state of the primary buffer.
 617  *
 618  * Returns:  None
 619  */
 620
 621 static void DSOUND_WaveQueue(DirectSoundDevice *device, LPBYTE pos, DWORD bytes)
 622 {
 623         BYTE *buffer;
 624         HRESULT hr;
 625
 626         TRACE("(%p)\n", device);
 627
 628         hr = IAudioRenderClient_GetBuffer(device->render, bytes / device->pwfx->nBlockAlign, &buffer);
 629         if(FAILED(hr)){
 630                 WARN("GetBuffer failed: %08x\n", hr);
 631                 return;
 632         }
 633
 634         memcpy(buffer, pos, bytes);
 635
 636         hr = IAudioRenderClient_ReleaseBuffer(device->render, bytes / device->pwfx->nBlockAlign, 0);
 637         if(FAILED(hr)) {
 638                 ERR("ReleaseBuffer failed: %08x\n", hr);
 639                 IAudioRenderClient_ReleaseBuffer(device->render, 0, 0);
 640                 return;
 641         }
 642
 643         device->pad += bytes;
 644 }
 645
 646 /**
 647  * Perform mixing for a Direct Sound device. That is, go through all the
 648  * secondary buffers (the sound bites currently playing) and mix them in
 649  * to the primary buffer (the device buffer).
 650  *
 651  * The mixing procedure goes:
 652  *
 653  * secondary->buffer (secondary format)
 654  *   =[Resample]=> device->tmp_buffer (float format)
 655  *   =[Volume]=> device->tmp_buffer (float format)
 656  *   =[Reformat]=> device->buffer (device format, skipped on float)
 657  */
 658 static void DSOUND_PerformMix(DirectSoundDevice *device)
 659 {
 660         DWORD block, pad_frames, pad_bytes, frames;
 661         HRESULT hr;
 662
 663         TRACE("(%p)\n", device);
 664
 665         /* **** */
 666         EnterCriticalSection(&device->mixlock);
 667
 668         hr = IAudioClient_GetCurrentPadding(device->client, &pad_frames);
 669         if(FAILED(hr)){
 670                 WARN("GetCurrentPadding failed: %08x\n", hr);
 671                 LeaveCriticalSection(&device->mixlock);
 672                 return;
 673         }
 674         block = device->pwfx->nBlockAlign;
 675         pad_bytes = pad_frames * block;
 676         device->playpos += device->pad - pad_bytes;
 677         device->playpos %= device->buflen;
 678         device->pad = pad_bytes;
 679
 680         frames = device->ac_frames - pad_frames;
 681         if(!frames){
 682                 /* nothing to do! */
 683                 LeaveCriticalSection(&device->mixlock);
 684                 return;
 685         }
 686         if (frames > device->frag_frames * 3)
 687                 frames = device->frag_frames * 3;
 688
 689         if (device->priolevel != DSSCL_WRITEPRIMARY) {
 690                 BOOL all_stopped = FALSE;
 691                 int nfiller;
 692                 void *buffer = NULL;
 693
 694                 /* the sound of silence */
 695                 nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
 696
 697                 /* check for underrun. underrun occurs when the write position passes the mix position
 698                  * also wipe out just-played sound data */
 699                 if (!pad_frames)
 700                         WARN("Probable buffer underrun\n");
 701
 702                 hr = IAudioRenderClient_GetBuffer(device->render, frames, (void*)&buffer);
 703                 if(FAILED(hr)){
 704                         WARN("GetBuffer failed: %08x\n", hr);
 705                         LeaveCriticalSection(&device->mixlock);
 706                         return;
 707                 }
 708
 709                 memset(buffer, nfiller, frames * block);
 710
 711                 if (!device->normfunction)
 712                         DSOUND_MixToPrimary(device, buffer, frames, &all_stopped);
 713                 else {
 714                         memset(device->buffer, nfiller, device->buflen);
 715
 716                         /* do the mixing */
 717                         DSOUND_MixToPrimary(device, (float*)device->buffer, frames, &all_stopped);
 718
 719                         device->normfunction(device->buffer, buffer, frames * device->pwfx->nChannels);
 720                 }
 721
 722                 hr = IAudioRenderClient_ReleaseBuffer(device->render, frames, 0);
 723                 if(FAILED(hr))
 724                         ERR("ReleaseBuffer failed: %08x\n", hr);
 725
 726                 device->pad += frames * block;
 727         } else if (!device->stopped) {
 728                 DWORD writepos = (device->playpos + pad_bytes) % device->buflen;
 729                 DWORD bytes = frames * block;
 730
 731                 if (bytes > device->buflen)
 732                         bytes = device->buflen;
 733                 if (writepos + bytes > device->buflen) {
 734                         DSOUND_WaveQueue(device, device->buffer + writepos, device->buflen - writepos);
 735                         DSOUND_WaveQueue(device, device->buffer, writepos + bytes - device->buflen);
 736                 } else
 737                         DSOUND_WaveQueue(device, device->buffer + writepos, bytes);
 738         }
 739
 740         LeaveCriticalSection(&(device->mixlock));
 741         /* **** */
 742 }
 743
 744 DWORD CALLBACK DSOUND_mixthread(void *p)
 745 {
 746         DirectSoundDevice *dev = p;
 747         TRACE("(%p)\n", dev);
 748
 749         while (dev->ref) {
 750                 DWORD ret;
 751
 752                 /*
 753                  * Some audio drivers are retarded and won't fire after being
 754                  * stopped, add a timeout to handle this.
 755                  */
 756                 ret = WaitForSingleObject(dev->sleepev, dev->sleeptime);
 757                 if (ret == WAIT_FAILED)
 758                         WARN("wait returned error %u %08x!\n", GetLastError(), GetLastError());
 759                 else if (ret != WAIT_OBJECT_0)
 760                         WARN("wait returned %08x!\n", ret);
 761                 if (!dev->ref)
 762                         break;
 763
 764                 RtlAcquireResourceShared(&(dev->buffer_list_lock), TRUE);
 765                 DSOUND_PerformMix(dev);
 766                 RtlReleaseResource(&(dev->buffer_list_lock));
 767         }
 768         SetEvent(dev->thread_finished);
 769         return 0;
 770 }